search for: res_features

...P/113-08674628 in 2/2 formats Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/113-08674628<ZOMBIE>' Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/113-08674628 (6) Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628<ZOMBIE> Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- Executing Set("SIP/111-086497c8", "DYNAMIC_FEATURES=") in new stack...

Hi, Gere are some messages that sometimes show up in my Asterisk logs... If you help me out to solve them, I could make a list of all know warning messages so that we can publish in the wiki or somewhere else! - "res_features.c: Did not read data." - on Google, the only reference to this was in Russian :( - "Asked to transmit frame type 64, while native formats is 256 (read/write = 256/256)" - I am using codecs g711 (for fax only) and g729 - "channel.c: Avoided deadlock for '0x87421a8', 10...

is where anyone who knows what is needed to get the pickupexten (*8) running ? gentoo asterisk-stable 1.2.4/zap1.2.4 with bristuff I've activated it in features.conf (default *8) and also tested other extensions res_features.so is loaded show features says: Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #1 Attended Transfer *2 One Touch Monitor *1 Disconnect Call *...

...ely cause > ill side effects or deadlocks on exit, so you need to play a bit > with modules.conf . > If you want to play with a very minimal version the following does something: > > ; -- modules.conf > [modules] > autoload=no > load => res_monitor.so > load => res_features.so > load => chan_sip.so > > Unfortunately, loading other modules is a bit critical and depending > on the order or the timing you get crashes etc. > > To build trunk under windows/cygwin you need at least the following pieces: > > bash > binutils > curl > g...

...ce a half year, i have done many updates of all packages and a clean install to merge this prob, no luck, it still exists. The facts i know about it : During such a " * DTMF Shooting" the logfiles recognized this (see the channel types!) : -- NOTICES -- Nov 6 09:53:26 WARNING[22637] res_features.c: Bridge failed on channels mISDN/1-1 and Zap/1-1 Nov 6 10:05:28 WARNING[22902] res_features.c: Bridge failed on channels SIP/40-0815e778 and SIP/pbx1-08281bc8 Nov 6 10:15:38 WARNING[23393] res_features.c: Bridge failed on channels SIP/40-0826c530 and IAX2/pbx1-1 DTMF Tone Log : Nov 6 05:00...

On my asterisk machines the following features.conf file crashes asterisk (core dump) This happens with 1.2.4, 1.2.10, 1.2.12, with or without bristuff. It's easy to work around, but broken nevertheless. Has anyone else experienced that or is it just me? ;) -------- /etc/asterisk/features.conf ---------------- [applicationmap] # THIS CRASHES asterisk: rateone => #1,caller,Macro,ratecall,1

...ting attended transfer with 3 SIP phones. I noticed about 10% of my transfers make the call drop and I get this on my log: Jul 5 10:42:32 WARNING[23960]: file.c:592 ast_readaudio_callback: Failed to write frame -- Playing 'beep' (language 'it') Jul 5 10:42:32 WARNING[23960]: res_features.c:745 builtin_atxfer: Failed to play transfer sound! Moreover, every time I try to transfer from called phone to a third phone I get this message: -- SIP/5-082a9f78 answered Local/12 at inbound_sip-f8de,2 Jul 5 13:02:40 NOTICE[24701]: res_features.c:1171 ast_feature_request_and_dial: Don&...

...] 's' => 1. Answer() [pbx_config] 2. Playback(vm-Work) [pbx_config] 3. Hangup() [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -= 2 extensions (4 priorities) in 3 contexts. =- ivr-server*CLI> El extension es este.... [general] language=en [from-ptsn] exten => s,1,Answer() exten => s,2,Playback(vm-Work) exten =&gt...

Asterisk don't running, because show this message WARNING[6949]: chan_sip.c:8865 reload_config: Section 'authentication' lacks type WARNING[6949]: chan_iax2.c:7491 load_module: Unable to open IAX timing interface: No such file or directory WARNING[6949]: chan_skinny.c:2587 reload_config: Unable to get our IP address, Skinny disabled WARNING[6949]: chan_oss.c:239 sound_thread: Read

...(*), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_CHANNEL_1 on c1 (SIP/rafal-89b1) Oct 17 15:52:23 DEBUG[20916]: channel.c:2798 ast_channel_bridge: Bridge stops bridging channels SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: res_features.c:600 ast_bridge_call: Read from SIP/rafal-89b1 (1,42) -- Attempting native bridge of SIP/andrzej-0265 and SIP/rafal-89b1 Oct 17 15:52:23 DEBUG[20916]: rtp.c:190 send_dtmf: Sending dtmf: 50 (2), at 10.2.20.65 Oct 17 15:52:23 DEBUG[20916]: channel.c:2762 ast_channel_bridge: Got AST_BRIDGE_DTMF_C...

...t's relevant, here's my modules.conf. Am I missing something important? [root@pbx asterisk]# cat modules.conf ; Modules.conf ; [modules] autoload=no ; Resources -- load => res_adsi.so ;load => res_agi.so ;load => res_config_odbc.so load => res_crypto.so load => res_features.so ;load => res_indications.so ;load => res_monitor.so load => res_musiconhold.so ;load => res_odbc.so ; PBX -- load => pbx_config.so ; Requires N/A ;load => pbx_dundi.so ; Requires res_crypto.so ;load => pbx_functions.so ; Requires N/A ;load => pbx_loopback.so ;...

...othing on the web about this, and I am not even sure where to start troubleshooting. Any help will be appreciated. Tommy Denton tommydenton@gmail.com Aug 14 10:06:38 VERBOSE[4326]: [res_adsi.so]Aug 14 10:06:38 VERBOSE[4326]: [res_adsi.so] => (ADSI Resource) Aug 14 10:06:38 VERBOSE[4326]: [res_features.so]Aug 14 10:06:38 VERBOSE[4326]: [res_features.so] => (Call Parking Resource) Aug 14 10:06:38 WARNING[4326]: Unknown directive 'parkingtime' at line 9 of features.conf Aug 14 10:06:38 VERBOSE[4326]: [res_indications.so]Aug 14 10:06:38 VERBOSE[4326]: [res_indications.so] => (Indica...

Hi All, I have a really strange issue occuring where if I run "show dialplan" or "dialplan show" or "dialplan show parkedcalls", then asterisk dumps core. It only appears to happen with contexts that are created within res_features. I am able to display all my other dialplans, but, every time I try to just do a normal "dialplan show" asterisk core dumps (Segmentation Fault). My environment is as follows: Asterisk v 1.4.11 Solaris 10 update 3 (11/06), patched current gcc v3.4.3 example console output -------------...

...6c4f98c, bridge_end= {tv_sec = 0, tv_usec = 0}) at channel.c:3260 #9 0x080655fd in ast_channel_bridge (c0=0xb659fcd0, c1=0x9455ca0, config=0xb6c4feb0, fo=0xb6c4f988, rc=0xb6c4f98c) at channel.c:3524 #10 0xb78fad29 in ast_bridge_call (chan=0xb659fcd0, peer=0x9455ca0, config=0xb6c4feb0) at res_features.c:1319 #11 0xb7099301 in dial_exec_full (chan=0xb659fcd0, data=0xb6c4feb0, peerflags=0xb6c50568) at app_dial.c:1577 #12 0xb7097dc5 in dial_exec (chan=0xb7ed1900, data=0xb7ed1900) at app_dial.c:1619 #13 0x0808e445 in pbx_extension_helper (c=0xb659fcd0, con=0xb7ed1900, context=0xb659fe20 "op05_x...

...everyone, I get these events sent like this: Event: ParkedCall Privilege: call,all Exten: 701 Channel: Zap/4-1 From: IAX2/cnw-4 Timeout: 120 CallerID: XXXXXXXXX CallerIDName: Conrad Wood Unqiueid: asterisk-1713-1139266402.909 ^^^^^^^^^ Please note the spelling of uniqueid. I find the spelling in res_features.c - but only once I patched it with bristuff patches. Does anyone know whether that is a known problem with bristuff? If so is it fixed in a later version? Where do I report a bug in bristuff? ;) Thanks, Conrad

...so tried to load manually: [Dec 18 14:31:26] WARNING[11002]: loader.c:359 load_dynamic_module: Error loadin ile: No such file or directory [Dec 18 14:31:26] WARNING[11002]: loader.c:653 load_resource: Module 'res_fax.so Verified the files exist: astbh00*CLI> module load res_f res_fax.so res_features.so res_fax_digium.so astbh00*CLI> module load res_f Help! :)

...3. Hangup() [pbx_config] Include => 'extentions' [pbx_config] Include => 'pasvalide' [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' => 1. Park() [res_features] -- Now, when I dial from any of the ext. to '0' It actually matches the '0', plays the goodbye message, but doesn't hangup but gets directly to the 'pasvalide' context. Sa...

...nfds=0, exception=0x0, outfd=0x0, ms=0xb7469b4c) at channel.c:1644 #3 0x08069d86 in ast_channel_bridge (c0=0xb22bf9a8, c1=0xa2ae648, config=0xb746a7a0, fo=0xb7469c40, rc=0xb7469c44) at channel.c:1721 #4 0x00548f65 in ast_bridge_call (chan=0xb22bf9a8, peer=0xa2ae648, config=0xb746a7a0) at res_features.c:1365 #5 0x005a40ba in dial_exec_full (chan=0xb22bf9a8, data=Variable "data" is not available. ) at app_dial.c:1633 #6 0x005a6a33 in dial_exec (chan=0xfffffffc, data=0x7fffffff) at app_dial.c:1680 #7 0x08090bad in pbx_extension_helper (c=0xb22bf9a8, con=Variable "con" is not...

...able extension, so that everyone else can just dial 5401, etc to pick up the call. I assume that there is some kind of local channel I can do this with and avoid any of the automated parking systems like Park, ParkandAnnounce or ValetParking, all of which have their pros and cons. Digging through res_features.c it is hard to discern where the call really is when parked (what kind of channel it is) so I am not sure if it can be reproduced in dialplan. Has anyone done manual call parking this way? Steven -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium...

Hi All, I read rumors of a potential Asterisk 1.4.12 release for last week. I would like to start testing Asterisk 1.4 on Solaris, but, the fix for the segfault in res_features is only in the current development trunk. I would much rather test a release version, and as such, need to wait for the release of 1.4.12, my question is, do we have a guestimate on when it will be released, 1 week, 2 weeks, a month? Thanks Bruce -------------- next part -------------- A non-text...